It is a problem in the field of data transmission that the rapid growth of the Internet and the transition of all multimedia content to digital formats have resulted in the use of packet-switched networks for the delivery of all video and audio content. Examples of such information delivery include, but are not limited to, delivery of voice services, music, and television entertainment over all forms of broadband data access networks, including: satellite, cable, DSL and cellular telephone networks using Internet Protocol (IP) based technology. The distribution of streaming media content over the Internet is simple in concept: the streaming media content is packetized into transport packets that are delivered from the content server to the user across the network as long as there is a persistent Internet Protocol connectivity between them.
The fundamental problem with this form of information delivery is that the Internet Protocol networks are not suited to support reliable delivery of continuous streams of data, since this network architecture is an economical “best efforts” packet delivery platform. For example, packets are discarded or dropped when the routers at the switching nodes in an Internet Protocol network cannot handle the traffic congestion caused by a surge in network traffic. Packet loss, which is an inherent property of the “best effort” nature of Internet Protocol networks, can cause serious degradations in streaming video quality.
The public Internet is an interconnection of packet-switched data communication networks, composed of a hybrid combination of various terrestrial wire-line and wireless transmission links, as well as satellite links. The Internet is poised to provide Internet Protocol connectivity for the delivery of many varieties of multimedia services and, in particular, video streaming services. High quality efficient video streaming transmission services are critical enablers for a variety of existing and emerging applications, such as Internet Protocol videoconferencing, live video entertainment, as well as on-demand video entertainment, including: Internet Protocol Television (IPTV), interactive gaming, telemedicine, remote teaching and training, and remote video surveillance. The delivery of streaming multimedia over the Internet combines regular audio and video content along with the synthetic multimedia content of virtual reality and the integrated imagery and graphics of web content to form a real-time interactive rich media event. The streaming multimedia content is delivered from the content source to the user in real-time. Live or pre-recorded content can be streamed according to a schedule and pushed from the content server to the user. Similarly, if the content has been stored for on-demand delivery, a streaming server delivers it at a controlled rate in real-time when the user requests and pulls down the content from the streaming server. This is in direct contrast to the legacy download-and-play process utilized to deliver streamed content in the early days of the Internet.
End-to-end streaming video quality can be impaired by packet losses either caused by dropped packets at routers due to network congestion or caused by corrupted packets over transmission links with excessive link noise or wireless propagation degradations (collectively termed “corrupted packets” herein). Streaming video often is acutely sensitive to even very low levels of packet loss. When interframe coding is used in video compression for delivery of digital TV, artifacts such as dropout, tiling, or pixelization caused by packet losses appearing in a reference video frame can impact all the frames within the reference frame picture group, which typically lasts between half a second to one second in duration. During channel change, compression decoders must wait until the next reference frame is assembled before presenting the new channel content to the user. Packet losses appearing in this frame can cause the decoder to have to wait until the next good reference frame, thus significantly increasing the TV channel change time.
Error control procedures can be applied at different layers of the Internet Protocol network protocol stack to handle packet losses. At the link level, Forward Error Correction Coding (FEC) can be employed in the Physical (PHY) layer to correct for bit errors in received packets caused by link impairments such as wireless channel noise and multi-path fading effects. In this approach, each packet is individually encoded with the Forward Error Correction Code to include some additional redundant bits that are transmitted along with the data bits. These redundant bits enable the decoder for the Forward Error Correction Code, which is located at the receiver, to correct for a number of bit errors in each packet within the error correction capability of the error correction code. Since Internet Protocol packets are discarded at the transport layer if its checksum fails (such as for packets with uncorrectable bit errors after the decoding process for the physical level Forward Error Correction Coding scheme is completed), only missing or lost packets can occur over the end-to-end network above the transport layer and up to the application layer.
Internet Protocol technology ensures end-to-end network packet delivery reliability through a scheme called Transmission Control Protocol (TCP) where missing or lost packets are retransmitted. Since real-time video streaming is an extremely delay-sensitive service, end-to-end retransmission methods such as that used in Transmission Control Protocol generally are not feasible except for direct transport over small local area networks with very short propagation delays. Instead, User Datagram Protocol (UDP) or Real Time Protocol (RTP) is commonly used in transporting real-time video streaming over Internet Protocol networks. Moreover, retransmission methods such as Transmission Control Protocol do not scale for multicast applications involving large user groups. Many Internet Protocol Television service providers have turned recently to the use of managed Internet Protocol networks in an attempt to address this packet loss issue. Managed networks prioritize real-time traffic, such as the video streaming packets, so that superior routing treatment is given to them. This can reduce the packet loss rate for streaming traffic. However the reduced level of packet loss rate still cannot ensure an end-to-end streaming video quality which matches that expected by legacy TV broadcast subscribers, let alone the more stringent quality of viewing experience expectations for High Definition TV.